System and method for digital signal processing

ABSTRACT

The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation-in-part of U.S. application Ser. No.14/059,948 which is a continuation-in-part of U.S. application Ser. No.12/648,007 filed on Dec. 28, 2009 which is a continuation-in-part ofU.S. application Ser. No. 11/947,301, filed Nov. 29, 2007, which claimspriority to U.S. Provisional Application No. 60/861,711 filed Nov. 30,2006, and is a continuation-in-part of U.S. application Ser. No.11/703,216, filed Feb. 7, 2007, which claims priority to U.S.Provisional Application No. 60/765,722, filed Feb. 7, 2006. Each of theabove applications is incorporated by reference herein in its entirety.

FIELD OF THE INVENTION

The present invention provides for methods and systems for digitallyprocessing an audio signal. Specifically, some embodiments relate todigitally processing an audio signal in order to deliver studio-qualitysound in a variety of consumer electronic devices.

BACKGROUND OF THE INVENTION

Historically, studio-quality sound, which can best be described as thefull reproduction of the complete range of audio frequencies that areutilized during the studio recording process, has only been able to beachieved, appropriately, in audio recording studios. Studio-qualitysound is characterized by the level of clarity and brightness which isattained only when the upper-mid frequency ranges are effectivelymanipulated and reproduced. While the technical underpinnings ofstudio-quality sound can be fully appreciated only by experienced recordproducers, the average listener can easily hear the difference thatstudio-quality sound makes.

While various attempts have been made to reproduce studio-quality soundoutside of the recording studio, those attempts have come at tremendousexpense (usually resulting from advanced speaker design, costlyhardware, and increased power amplification) and have achieved onlymixed results. Thus, there exists a need for a process wherebystudio-quality sound can be reproduced outside of the studio withconsistent, high quality results at a low cost. There exists a furtherneed for audio devices embodying such a process in the form of computerchips embedded within audio devices, or within processing devicesseparate and standalone from the audio devices. There also exists a needfor the ability to produce studio-quality sound through inexpensivespeakers, as well as through a variety of readily available consumerdevices capable of reproducing sound, in both hardware-based andsoftware-based embodiments.

SUMMARY OF THE INVENTION

The present invention meets the existing needs described above byproviding for a system and method of digitally processing an audiosignal in a manner such that studio-quality sound can be reproducedacross the entire spectrum of audio devices. The present invention alsoprovides for the ability to enhance audio in real-time and tailors theenhancement to the audio signal of a given audio device or deliverysystem and playback environment.

The present invention may provide for a computer chip that can digitallyprocess an audio signal in such a manner, as well as provide for audiodevices that comprise such a chip or equivalent circuit combination. Thepresent invention may also provide for computer software readable andexecutable by a computer to digitally process an audio signal. In thesoftware embodiments, the present invention may utilize existinghardware and software components on computers such as PCs, Mac, andmobile devices, comprising various operating systems such as Android,iOS, and Windows.

Accordingly, in initially broad terms, an audio input signal is firstfiltered with a high pass filter. The high pass filter, in at least oneembodiment, is configured to remove ultra-low frequency content from theinput audio signal resulting in the generation of a high pass signal.

The high pass signal from the high pass filter is then filtered througha first filter module to create a first filtered signal. The firstfilter module is configured to selectively boost and/or attenuate thegain of select frequency ranges in an audio signal, such as the highpass signal. In at least one embodiment, the first filter module boostsfrequencies above a first frequency, and attenuates frequencies below afirst frequency.

The first filtered signal from the first filter module is then modulatedwith a first compressor to create a modulated signal. The firstcompressor is configured for the dynamic range compression of a signal,such as the first filtered signal. Because the first filtered signalboosted higher frequencies and attenuated lower frequencies, the firstcompressor may, in at least one embodiment, be configured to trigger andadjust the higher frequency material, while remaining relativelyinsensitive to lower frequency material.

The modulated signal from the first compressor is then filtered througha second filter module to create a second filtered signal. The secondfilter module is configured to selectively boost and/or attenuate thegain of select frequency ranges in an audio signal, such as themodulated signal. In at least one embodiment, the second filter moduleis configured to be of least partially a inverse relation relative tothe first filter module. For example, if the first filter module boostedcontent above a first frequency by +X dB and attenuated content below afirst frequency by −Y dB, the second filter module may then attenuatethe content above the first frequency by −X dB, and boost the contentbelow the first frequency by +Y dB. In other words, the purpose of thesecond filter module in one embodiment may be to “undo” the gainadjustment that was applied by the first filter module.

The second filtered signal from the second filter module is thenprocessed with a first processing module to create a processed signal.In at least one embodiment, the first processing module may comprise apeak/dip module. In other embodiments, the first processing module maycomprise both a peak/dip module and a first gain element. The first gainelement may be configured to adjust the gain of the signal, such as thesecond filtered signal. The peak/dip module may be configured to shapethe signal, such as to increase or decrease overshoots or undershoots inthe signal.

The processed signal from the first processing module is then split witha band splitter into a low band signal, a mid band signal and a highband signal. In at least one embodiment, each band may comprise theoutput of a fourth order section, which may be realized as the cascadeof second order biquad filters.

The low band signal is modulated with a low band compressor to create amodulated low band signal, and the high band signal is modulated with ahigh band compressor to create a modulated high band signal. The lowband compressor and high band compressor are each configured todynamically adjust the gain of a signal. Each of the low band compressorand high band compressor may be computationally and/or configuredidentically as the first compressor.

The modulated low band signal, the mid band signal, and the modulatedhigh band signal are then processed with a second processing module. Thesecond processing module may comprise a summing module configured tocombine the signals. The summing module in at least one embodiment mayindividually alter the gain of each of the modulated low band, mid band,and modulated high band signals. The second processing module mayfurther comprise a second gain element. The second gain element mayadjust the gain of the combined signal in order to create an outputsignal.

These and other objects, features and advantages of the presentinvention will become clearer when the drawings as well as the detaileddescription are taken into consideration.

BRIEF DESCRIPTION OF THE DRAWINGS

For a fuller understanding of the nature of the present invention,reference should be had to the following detailed description taken inconnection with the accompanying drawings in which:

FIG. 1 illustrates a schematic of one embodiment of the presentinvention directed to a system for digitally processing an audio signal.

FIG. 2 illustrates a schematic of another embodiment of the presentinvention directed to a system for digitally processing an audio signal.

FIG. 3 illustrates a block diagram of another embodiment of the presentinvention directed to a method for digitally processing an audio signal.

FIG. 4 illustrates a block diagram of another embodiment of the presentinvention directed to a method for digitally processing an audio signal.

Like reference numerals refer to like parts throughout the several viewsof the drawings.

DETAILED DESCRIPTION OF THE EMBODIMENT

As illustrated by the accompanying drawings, the present invention isdirected to systems and methods for digitally processing an audiosignal. Specifically, some embodiments relate to digitally processing anaudio signal in order to deliver studio-quality sound in a variety ofdifferent consumer electronic devices.

As schematically represented, FIG. 1 illustrates at least one preferredembodiment of a system 100 for digitally processing an audio signal, andFIG. 2 provides examples of several subcomponents and combinations ofsubcomponents of the modules of FIG. 1. Accordingly, and in theseembodiments, the systems 100 and 300 generally comprise an input device101, a high pass filter 111, a first filter module 301, a firstcompressor 114, a second filter module 302, a first processing module303, a band splitter 119, a low band compressor 130, a high bandcompressor 131, a second processing module 304, and an output device102.

The input device 101 is at least partially structured or configured totransmit an input audio signal 201 into the system 100 of the presentinvention, and in at least one embodiment into the high pass filter 111.The input audio signal 201 may comprise the full audible range, orportions of the audible range. The input audio signal 201 may comprise astereo audio signal. The input device 101 may comprise at least portionsof an audio device capable of audio playback. The input device 101 forinstance, may comprise a stereo system, a portable music player, amobile device, a computer, a sound or audio card, or any other device orcombination of electronic circuits suitable for audio playback.

The high pass filter 111 is configured to pass through high frequenciesof an audio signal, such as the input signal 201, while attenuatinglower frequencies, based on a predetermined frequency. In other words,the frequencies above the predetermined frequency may be transmitted tothe first filter module 301 in accordance with the present invention. Inat least one embodiment, ultra-low frequency content is removed from theinput audio signal, where the predetermined frequency may be selectedfrom a range between 300 Hz and 3 kHz. The predetermined frequencyhowever, may vary depending on the source signal, and vary in otherembodiments to comprise any frequency selected from the full audiblerange of frequencies between 20 Hz to 20 kHz. The predeterminedfrequency may be tunable by a user, or alternatively be statically set.The high pass filter 111 may further comprise any circuits orcombinations thereof structured to pass through high frequencies above apredetermined frequency, and attenuate or filter out the lowerfrequencies.

The first filter module 301 is configured to selectively boost orattenuate the gain of select frequency ranges within an audio signal,such as the high pass signal 211. For example, and in at least oneembodiment, frequencies below a first frequency may be adjusted by ±XdB, while frequencies above a first frequency may be adjusted by ±Y dB.In other embodiments, a plurality of frequencies may be used toselectively adjust the gain of various frequency ranges within an audiosignal. In at least one embodiment, the first filter module 301 may beimplemented with a first low shelf filter 112 and a first high shelffilter 113, as illustrated in FIG. 1. The first low shelf filter 112 andfirst high shelf filter 113 may both be second-order filters. In atleast one embodiment, the first low shelf filter 112 attenuates contentbelow a first frequency, and the first high shelf filter 112 boostscontent above a first frequency. In other embodiments, the frequencyused for the first low shelf filter 112 and first high shelf filter 112may comprise two different frequencies. The frequencies may be static oradjustable. Similarly, the gain adjustment (boost or attenuation) may bestatic or adjustable.

The first compressor 114 is configured to modulate a signal, such as thefirst filtered signal 401. The first compressor 112 may comprise anautomatic gain controller. The first compressor 112 may comprisestandard dynamic range compression controls such as threshold, ratio,attack and release. Threshold allows the first compressor 112 to reducethe level of the filtered signal 211 if its amplitude exceeds a certainthreshold. Ratio allows the first compressor 112 to reduce the gain asdetermined by a ratio. Attack and release determines how quickly thefirst compressor 112 acts. The attack phase is the period when the firstcompressor 112 is decreasing gain to reach the level that is determinedby the threshold. The release phase is the period that the firstcompressor 112 is increasing gain to the level determined by the ratio.The first compressor 112 may also feature soft and hard knees to controlthe bend in the response curve of the output or modulated signal 212,and other dynamic range compression controls appropriate for the dynamiccompression of an audio signal. The first compressor 112 may furthercomprise any device or combination of circuits that is structured andconfigured for dynamic range compression.

The second filter module 302 is configured to selectively boost orattenuate the gain of select frequency ranges within an audio signal,such as the modulated signal 214. In at least one embodiment, the secondfilter module 302 is of the same configuration as the first filtermodule 301. Specifically, the second filter module 302 may comprise asecond low shelf filter 115 and a second high shelf filter 116. Thesecond filter module 302 may be configured in at least a partiallyinverse configuration to the first filter module 301. For instance, thesecond filter module may use the same frequency, for instance the firstfrequency, as the first filter module. Further, the second filter modulemay adjust the gain inversely to the gain or attenuation of the firstfilter module, of content above the first frequency. Similarly secondfilter module may also adjust the gain inversely to the gain orattenuation of the of the first filter module, of content below thefirst frequency. In other words, the purpose of the second filter modulein one embodiment may be to “undo” the gain adjustment that was appliedby the first filter module.

The first processing module 303 is configured to process a signal, suchas the second filtered signal 402. In at least one embodiment, the firstprocessing module 303 may comprise a peak/dip module, such as 118represented in FIG. 2. In other embodiments, the first processing module303 may comprise a first gain element 117. In various embodiments, theprocessing module 303 may comprise both a first gain element 117 and apeak/dip module 118 for the processing of a signal. The first gainelement 117, in at least one embodiment, may be configured to adjust thelevel of a signal by a static amount. The first gain element 17 maycomprise an amplifier or a multiplier circuit. In other embodiments,dynamic gain elements may be used. The peak/dip module 118 is configuredto shape the desired output spectrum, such as to increase or decreaseovershoots or undershoots in the signal. In some embodiments, thepeak/dip module may further be configured to adjust the slope of asignal, for instance for a gradual scope that gives a smoother response,or alternatively provide for a steeper slope for more sudden sounds. Inat least one embodiment, the peak/dip module 118 comprises a bank of tencascaded peak/dipping filters. The bank of ten cascaded peaking/dippingfilters may further be second-order filters. In at least one embodiment,the peak/dip module 118 may comprise an equalizer, such as parametric orgraphic equalizers.

The band splitter 119 is configured to split a signal, such as theprocessed signal 403. In at least one embodiment, the signal is splitinto a low band signal 220, a mid band signal 221, and a high bandsignal 222. Each band may be the output of a fourth order section, whichmay be further realized as the cascade of second order biquad filters.In other embodiments, the band splitter may comprise any combination ofcircuits appropriate for splitting a signal into three frequency bands.The low, mid, and high bands may be predetermined ranges, or may bedynamically determined based on the frequency itself, i.e. a signal maybe split into three even frequency bands, or by percentage. Thedifferent bands may further be defined or configured by a user and/orcontrol mechanism.

A low band compressor 130 is configured to modulate the low band signal220, and a high band compressor 131 is configured to modulate the highband signal 222. In at least one embodiment, each of the low bandcompressor 130 and high band compressor 131 may be the same as the firstcompressor 114. Accordingly, each of the low band compressor 130 andhigh band compressor 131 may each be configured to modulate a signal.Each of the compressors 130, 131 may comprise an automatic gaincontroller, or any combination of circuits appropriate for the dynamicrange compression of an audio signal.

A second processing module 304 is configured to process at least onesignal, such as the modulated low band signal 230, the mid band signal221, and the modulated high band signal 231. Accordingly, the secondprocessing module 304 may comprise a summing module 132 configured tocombine a plurality of signals. The summing module 132 may comprise amixer structured to combine two or more signals into a composite signal.The summing module 132 may comprise any circuits or combination thereofstructured or configured to combine two or more signals. In at least oneembodiment, the summing module 132 comprises individual gain controlsfor each of the incoming signals, such as the modulated low band signal230, the mid band signal 221, and the modulated high band signal 231. Inat least one embodiment, the second processing module 304 may furthercomprise a second gain element 133. The second gain element 133, in atleast one embodiment, may be the same as the first gain element 117. Thesecond gain element 133 may thus comprise an amplifier or multipliercircuit to adjust the signal, such as the combined signal, by apredetermined amount.

The output device 102 may be structured to further process the outputsignal 404. The output device 102 may also be structured and/orconfigured for playback of the output signal 404.

As diagrammatically represented, FIG. 3 illustrates another embodimentdirected to a method for digitally processing an audio signal, which mayin at least one embodiment incorporate the components or combinationsthereof from the systems 100 and/or 300 referenced above. Each step ofthe method in FIG. 3 as detailed below may also be in the form of a codesegment directed to at least one embodiment of the present invention,which is stored on a non-transitory computer readable medium, forexecution by a computer to process an input audio signal.

Accordingly, an input audio signal is first filtered, as in 501, with ahigh pass filter to create a high pass signal. The high pass filter isconfigured to pass through high frequencies of a signal, such as theinput signal, while attenuating lower frequencies. In at least oneembodiment, ultra-low frequency content is removed by the high-passfilter. In at least one embodiment, the high pass filter may comprise afourth-order filter realized as the cascade of two second-order biquadsections. The reason for using a fourth order filter broken into twosecond order sections is that it allows the filter to retain numericalprecision in the presence of finite word length effects, which canhappen in both fixed and floating point implementations. An exampleimplementation of such an embodiment may assume a form similar to thefollowing:

-   -   Two memory locations are allocated, designated as d(k−1) and        d(k−2), with each holding a quantity known as a state variable.        For each input sample x(k), a quantity d(k) is calculated using        the coefficients a1 and a2:

d(k)=x(k)−a1*d(k−1)−a2*d(k−2)

-   -   -   The output y(k) is then computed, based on coefficients b0,            b1, and b2, according to:

y(k)=b0*d(k)+b1*d(k−1)+b2*d(k−2)

The above computation comprising five multiplies and four adds isappropriate for a single channel of second-order biquad section.Accordingly, because the fourth-order high pass filter is realized as acascade of two second-order biquad sections, a single channel of fourthorder input high pass filter would require ten multiples, four memorylocations, and eight adds.

The high pass signal from the high pass filter is then filtered, as in502, with a first filter module to create a first filtered signal. Thefirst filter module is configured to selectively boost or attenuate thegain of select frequency ranges within an audio signal, such as the highpass signal. Accordingly, the first filter module may comprise a secondorder low shelf filter and a second order high shelf filter in at leastone embodiment. In at least one embodiment, the first filter moduleboosts the content above a first frequency by a certain amount, andattenuates the content below a first frequency by a certain amount,before presenting the signal to a compressor or dynamic rangecontroller. This allows the dynamic range controller to trigger andadjust higher frequency material, whereas it is relatively insensitiveto lower frequency material.

The first filtered signal from the first filter module is thenmodulated, as in 503, with a first compressor. The first compressor maycomprise an automatic or dynamic gain controller, or any circuitsappropriate for the dynamic compression of an audio signal. Accordingly,the compressor may comprise standard dynamic range compression controlssuch as threshold, ratio, attack and release. An example implementationof the first compressor may assume a form similar to the following:

-   -   The compressor first computes an approximation of the signal        level, where att represents attack time; rel represents release        time; and invThr represents a precomputed threshold:

temp=abs(x(k))

if temp>level(k−1)

level(k)=att*(level(k−1)−temp)+temp

else

level=rel*(level(k−1)−temp)+temp

This level computation is done for each input sample. The ratio of thesignal's level to invThr then determines the next step. If the ratio isless than one, the signal is passed through unaltered. If the ratioexceeds one, a table in the memory may provide a constant that's afunction of both invThr and level:

if (level*thr<1)

output(k)=x(k)

else

index=floor(level*invThr)

if (index>99)

index=99

gainReduction=table[index]

output(k)=gainReduction*x(k)

The modulated signal from the first compressor is then filtered, as in504, with a second filter module to create a second filtered signal. Thesecond filter module is configured to selectively boost or attenuate thegain of select frequency ranges within an audio signal, such as themodulated signal. Accordingly, the second filter module may comprise asecond order low shelf filter and a second order high shelf filter in atleast one embodiment. In at least one embodiment, the second filtermodule boosts the content above a second frequency by a certain amount,and attenuates the content below a second frequency by a certain amount.In at least one embodiment, the second filter module adjusts the contentbelow the first specified frequency by a fixed amount, inverse to theamount that was removed by the first filter module. By way of example,if the first filter module boosted content above a first frequency by +XdB and attenuated content below a first frequency by −Y dB, the secondfilter module may then attenuate the content above the first frequencyby −X dB, and boost the content below the first frequency by +Y dB. Inother words, the purpose of the second filter module in one embodimentmay be to “undo” the filtering that was applied by the first filtermodule.

The second filtered signal from the second filter module is thenprocessed, as in 505, with a first processing module to create aprocessed signal. The processing module may comprise a gain elementconfigured to adjust the level of the signal. This adjustment, forinstance, may be necessary because the peak-to-average ratio wasmodified by the first compressor. The processing module may comprise apeak/dip module. The peak/dip module may comprise ten cascadedsecond-order filters in at least one embodiment. The peak/dip module maybe used to shape the desired output spectrum of the signal. In at leastone embodiment, the first processing module comprises only the peak/dipmodule. In other embodiments, the first processing module comprises again element followed by a peak/dip module.

The processed signal from the first processing module is then split, asin 506, with a band splitter into a low band signal, a mid band signal,and a high band signal. The band splitter may comprise any circuit orcombination of circuits appropriate for splitting a signal into aplurality of signals of different frequency ranges. In at least oneembodiment, the band splitter comprises a fourth-order band-splittingbank. In this embodiment, each of the low band, mid band, and high bandare yielded as the output of a fourth-order section, realized as thecascade of second-order biquad filters.

The low band signal is modulated, as in 507, with a low band compressorto create a modulated low band signal. The low band compressor may beconfigured and/or computationally identical to the first compressor inat least one embodiment. The high band signal is modulated, as in 508,with a high band compressor to create a modulated high band signal. Thehigh band compressor may be configured and/or computationally identicalto the first compressor in at least one embodiment.

The modulated low band signal, mid band signal, and modulated high bandsignal are then processed, as in 509, with a second processing module.The second processing module comprises at least a summing module. Thesumming module is configured to combine a plurality of signals into onecomposite signal. In at least one embodiment, the summing module mayfurther comprise individual gain controls for each of the incomingsignals, such as the modulated low band signal, the mid band signal, andthe modulated high band signal. By way of example, an output of thesumming module may be calculated by:

out=w0*low+w1*mid+w2*high

The coefficients w0, w1, and w2 represent different gain adjustments.The second processing module may further comprise a second gain element.The second gain element may be the same as the first gain element in atleast one embodiment. The second gain element may provide a final gainadjustment. Finally, the second processed signal is transmitted as theoutput signal.

As diagrammatically represented, FIG. 4 illustrates another embodimentdirected to a method for digitally processing an audio signal, which mayin at least one embodiment incorporate the components or combinationsthereof from the systems 100 and/or 300 referenced above. Because theindividual components of FIG. 4 have been discussed in detail above,they will not be discussed here. Further, each step of the method inFIG. 4 as detailed below may also be in the form of a code segmentdirected to at least one embodiment of the present invention, which isstored on a non-transitory computer readable medium, for execution by acomputer to process an input audio signal.

Accordingly, an input audio signal is first filtered, as in 501, with ahigh pass filter. The high pass signal from the high pass filter is thenfiltered, as in 601, with a first low shelf filter. The signal from thefirst low shelf filter is then filtered with a first high shelf filter,as in 602. The first filtered signal from the first low shelf filter isthen modulated with a first compressor, as in 503. The modulated signalfrom the first compressor is filtered with a second low shelf filter asin 611. The signal from the low shelf filter is then filtered with asecond high shelf filter, as in 612. The second filtered signal from thesecond low shelf filter is then gain-adjusted with a first gain element,as in 621. The signal from the first gain element is further processedwith a peak/dip module, as in 622. The processed signal from thepeak/dip module is then split into a low band signal, a mid band signal,and a high band signal, as in 506. The low band signal is modulated witha low band compressor, as in 507. The high band signal is modulated witha high band compressor, as in 508. The modulated low band signal, midband signal, and modulated high band signal are then combined with asumming module, as in 631. The combined signal is then gain adjustedwith a second gain element in order to create the output signal, as in632.

Any of the above methods may be completed in sequential order in atleast one embodiment, though they may be completed in any other order.In at least one embodiment, the above methods may be exclusivelyperformed, but in other embodiments, one or more steps of the methods asdescribed may be skipped.

Since many modifications, variations and changes in detail can be madeto the described preferred embodiment of the invention, it is intendedthat all matters in the foregoing description and shown in theaccompanying drawings be interpreted as illustrative and not in alimiting sense. Thus, the scope of the invention should be determined bythe appended claims and their legal equivalents.

Now that the invention has been described,

What is claimed is:
 1. A system for digital signal processing of aninput audio signal comprising: a high pass filter configured to filterthe input audio signal to create a high pass signal, a first low shelffilter configured to filter the high pass signal to create a first lowshelf signal, a first high shelf filter configured to filter the firstlow shelf signal to create the first filtered signal. a first compressorconfigured to modulate the first filtered signal to create a modulatedsignal, a second low shelf filter configured to between 100 Hz to 3000Hz and −5 dB to −20 dB to filter the modulated signal to create a secondlow shelf signal, a second high shelf filter configured to between 100Hz to 3000 Hz and +5 dB to +20 dB to filter the second low shelf signalto create the second filtered signal, a first processing moduleconfigured to process second filtered signal to create a processedsignal, a band splitter configured to split the processed signal into alow band signal, a mid band signal, and a high band signal, a low bandcompressor configured to modulate the low band signal to create amodulated low band signal, a high band compressor configured to modulatethe high band signal to create a modulated high band signal, and asecond processing module configured to process the modulated low bandsignal, the mid band signal, and the modulated high band signal tocreate an output signal.
 2. A system as recited in claim 1 wherein saidsecond low shelf filter and said second high shelf filter are structuredto establish a 24 dB differential between high and low frequencies forthe second filtered signal.
 3. A system as recited in claim 2 whereinsaid second low shelf filter is set to 1000 Hz and −12 dB and saidsecond high shelf filter is set to the 1000 Hz and −24 dB.
 4. A systemas recited in claim 1 wherein said first low shelf filter is set to 500Hz and −24 dB and said second high shelf filter is set to the 200 Hz and−24 dB
 5. A system as recited in claim 1 wherein said processing modulecomprises a peak/dip module configured to process the second filteredsignal to create the processed signal.
 6. A system as recited in claim 1wherein said first processing module comprises: a first gain elementconfigured to adjust the gain of the second filtered signal to create afirst gain signal, a peak/dip module configured to process the firstgain signal to create the processed signal.
 7. A system as recited inclaim 1 wherein said second processing module comprises a summing moduleconfigured to combine the modulated low band signal, the mid bandsignal, and the modulated high band signal to create the output signal.8. A system as recited in claim 1 wherein said second processing modulecomprises: a summing module configured to combine the modulated low bandsignal, the mid band signal, and the modulated high band signal tocreate a combined signal, a second gain element configured to adjust thegain of the combined signal to create the output signal.
 9. A system asrecited in claim 1 wherein said high pass filter comprises a fourthorder high pass filter.
 10. A system as recited in claim 1 wherein saidfirst low shelf filter comprises a second order low shelf filter.
 11. Asystem as recited in claim 1 wherein said first high shelf filtercomprises a second order high shelf filter.
 12. A system as recited inclaim 1 wherein said second low shelf filter comprises a second orderlow shelf filter.
 13. A system as recited in claim 1 wherein said secondhigh shelf filter comprises a second order high shelf filter.